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Fixed audio resampling to now use JUCE instead of FFmpeg. Removed lots of debug code.

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This commit is contained in:
Jonathan Thomas
2012-07-30 11:05:36 -05:00
parent 0aac7871a3
commit 80ca5e8315
2 changed files with 15 additions and 92 deletions
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57
src/FFmpegWriter.cpp
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@@ -524,17 +524,11 @@ void FFmpegWriter::write_audio_packet(Frame* frame)
{
// Translate sample value and copy into buffer
frame_samples[s] = int(frame_samples_float[s] * (1 << 15));
// // DEBUG
// if (frame->number == 1)
// {
// cout << frame_samples[s] << endl;
// }
}
// Re-sample audio samples (if needed)
if (c->sample_fmt != AV_SAMPLE_FMT_S16 || info.channels != channels_in_frame || info.sample_rate != sample_rate_in_frame) {
// Re-sample audio samples (into additinal channels or changing the sample format / number format)
// The sample rate has already been resampled using the GetInterleavedAudioSamples method.
if (c->sample_fmt != AV_SAMPLE_FMT_S16 || info.channels != channels_in_frame) {
// Audio needs to be converted
// Create an audio resample context object (used to convert audio samples)
@@ -550,14 +544,6 @@ void FFmpegWriter::write_audio_packet(Frame* frame)
total_frame_samples = audio_resample(resampleCtx, (short *) converted_audio, (short *) frame_samples, total_frame_samples);
remaining_frame_samples = total_frame_samples;
// DEBUG
// cout << endl << endl << endl << "END OF RESAMPLE DATA:" << endl;
// if (frame->number == 1)
// {
// for (int s = total_frame_samples - 10; s < (total_frame_samples + 200); s++)
// cout << converted_audio[s] << endl;
// }
// Copy audio samples over original samples
memcpy(frame_samples, converted_audio, total_frame_samples * av_get_bytes_per_sample(c->sample_fmt));
@@ -566,14 +552,6 @@ void FFmpegWriter::write_audio_packet(Frame* frame)
}
}
// // DEBUG
// cout << endl << endl << endl << "AFTER RESCALE" << endl;
// if (frame->number == 1)
// {
// for (int s = 0; s < total_frame_samples; s++)
// cout << frame_samples[s] << endl;
// }
// Loop until no more samples
while (remaining_frame_samples > 0) {
// Get remaining samples needed for this packet
@@ -586,13 +564,6 @@ void FFmpegWriter::write_audio_packet(Frame* frame)
else if (remaining_frame_samples < remaining_packet_samples)
diff = remaining_frame_samples;
// DEBUG
cout << "Frame: " << frame->number << ", audio_input_position: " << audio_input_position << ", samples_position: " << samples_position << ", total_frame_samples: " << total_frame_samples << ", audio_input_frame_size: " << audio_input_frame_size << endl;
cout << " -- remaining_frame_samples: " << remaining_frame_samples << ", remaining_packet_samples: " << remaining_packet_samples << endl;
cout << " -- Wrote " << diff << endl;
// Copy samples into input buffer (and convert to 16 bit int)
for (int s = 0; s < diff; s++)
{
@@ -604,33 +575,13 @@ void FFmpegWriter::write_audio_packet(Frame* frame)
samples_position++;
remaining_frame_samples--;
remaining_packet_samples--;
// DEBUG
// if (frame->number == 1 && s > 2800)
// {
// cout << frame_samples[s] << endl;
// }
// if (frame->number == 2 && s < 100)
// {
// cout << frame_samples[s] << endl;
// }
}
// if (frame->number == 2 || frame->number == 3)
// {
// for (int z = 0; z < diff; z++)
// cout << frame_samples[z] << endl;
// }
// Do we have enough samples to proceed?
if (audio_input_position < audio_input_frame_size)
// Not enough samples to encode... so wait until the next frame
break;
cout << "NOW ENCODE FILE!" << endl;
// Set packet properties
pkt.size = avcodec_encode_audio(c, audio_outbuf, audio_outbuf_size, (short *) samples);
@@ -640,8 +591,6 @@ void FFmpegWriter::write_audio_packet(Frame* frame)
pkt.stream_index = audio_st->index;
pkt.data = audio_outbuf;
cout << "Final encode: frame " << frame->number << ", pkt.pts: " << pkt.pts << endl;
/* write the compressed frame in the media file */
if (av_interleaved_write_frame(oc, &pkt) != 0)
throw ErrorEncodingAudio("Error while writing audio frame", frame->number);